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Video d’abeille en portrait
14 mai 2011, par
Mis à jour : Février 2012
Langue : français
Type : Video
Autres articles (26)
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Contribute to a better visual interface
13 avril 2011MediaSPIP is based on a system of themes and templates. Templates define the placement of information on the page, and can be adapted to a wide range of uses. Themes define the overall graphic appearance of the site.
Anyone can submit a new graphic theme or template and make it available to the MediaSPIP community. -
Ajouter notes et légendes aux images
7 février 2011, parPour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
Modification lors de l’ajout d’un média
Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...) -
Support de tous types de médias
10 avril 2011Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)
Sur d’autres sites (6797)
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Revision 38fa487164 : Shortcut 8x8/16x16 inverse 2D-DCT This commit brought back the shortcut impleme
27 juillet 2013, par Jingning HanChanged Paths :
Modify /vp9/decoder/vp9_idct_blk.c
Modify /vp9/encoder/vp9_encodemb.c
Shortcut 8x8/16x16 inverse 2D-DCTThis commit brought back the shortcut implementation of 8x8/16x16
inverse 2D-DCT. When the eob <= 10, it skips the inverse transform
operations on row 4:7/4:15 in the first round. For bus_cif at 1000
kbps, this provides about 2% speed-up at speed 0.Change-Id : I453e2d72956467d75be4ad8c04b4482ab889d572
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How to concat mp4 files using libffmpeg in c program ?
1er août 2013, par chichienI know ffmpeg command line is easy, but how to programmatically implement？ I'm not good at this，here is some code from internet, it is used to convert .mp4 to .ts，and i made some changes，but the audio stream problem persists：
#include
#include
#include
#include
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libavutil/avutil.h"
#include "libavutil/rational.h"
#include "libavdevice/avdevice.h"
#include "libavutil/mathematics.h"
#include "libswscale/swscale.h"
static AVStream* add_output_stream(AVFormatContext* output_format_context, AVStream* input_stream)
{
AVCodecContext* input_codec_context = NULL;
AVCodecContext* output_codec_context = NULL;
AVStream* output_stream = NULL;
output_stream = av_new_stream(output_format_context, 0);
if (!output_stream)
{
printf("Call av_new_stream function failed\n");
return NULL;
}
input_codec_context = input_stream->codec;
output_codec_context = output_stream->codec;
output_codec_context->codec_id = input_codec_context->codec_id;
output_codec_context->codec_type = input_codec_context->codec_type;
output_codec_context->codec_tag = input_codec_context->codec_tag;
output_codec_context->bit_rate = input_codec_context->bit_rate;
output_codec_context->extradata = input_codec_context->extradata;
output_codec_context->extradata_size = input_codec_context->extradata_size;
if (av_q2d(input_codec_context->time_base) * input_codec_context->ticks_per_frame > av_q2d(input_stream->time_base) && av_q2d(input_stream->time_base) < 1.0 / 1000)
{
output_codec_context->time_base = input_codec_context->time_base;
output_codec_context->time_base.num *= input_codec_context->ticks_per_frame;
}
else
{
output_codec_context->time_base = input_stream->time_base;
}
switch (input_codec_context->codec_type)
{
case AVMEDIA_TYPE_AUDIO:
output_codec_context->channel_layout = input_codec_context->channel_layout;
output_codec_context->sample_rate = input_codec_context->sample_rate;
output_codec_context->channels = input_codec_context->channels;
output_codec_context->frame_size = input_codec_context->frame_size;
if ((input_codec_context->block_align == 1 && input_codec_context->codec_id == CODEC_ID_MP3) || input_codec_context->codec_id == CODEC_ID_AC3)
{
output_codec_context->block_align = 0;
}
else
{
output_codec_context->block_align = input_codec_context->block_align;
}
break;
case AVMEDIA_TYPE_VIDEO:
output_codec_context->pix_fmt = input_codec_context->pix_fmt;
output_codec_context->width = input_codec_context->width;
output_codec_context->height = input_codec_context->height;
output_codec_context->has_b_frames = input_codec_context->has_b_frames;
if (output_format_context->oformat->flags & AVFMT_GLOBALHEADER)
{
output_codec_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
break;
default:
break;
}
return output_stream;
}
//[[** from ffmpeg.c
static void write_frame(AVFormatContext *s, AVPacket *pkt, AVCodecContext *avctx, AVBitStreamFilterContext *bsfc){
int ret;
while(bsfc){
AVPacket new_pkt= *pkt;
int a= av_bitstream_filter_filter(bsfc, avctx, NULL,
&new_pkt.data, &new_pkt.size,
pkt->data, pkt->size,
pkt->flags & AV_PKT_FLAG_KEY);
if(a>0){
av_free_packet(pkt);
new_pkt.destruct= av_destruct_packet;
} else if(a<0){
fprintf(stderr, "%s failed for stream %d, codec %s\n",
bsfc->filter->name, pkt->stream_index,
avctx->codec ? avctx->codec->name : "copy");
//print_error("", a);
//if (exit_on_error)
// ffmpeg_exit(1);
}
*pkt= new_pkt;
bsfc= bsfc->next;
}
ret= av_interleaved_write_frame(s, pkt);
if(ret < 0){
//print_error("av_interleaved_write_frame()", ret);
fprintf(stderr, "av_interleaved_write_frame(%d)\n", ret);
exit(1);
}
}
//]]**
int main(int argc, char* argv[])
{
const char* input;
const char* output;
const char* output_prefix = NULL;
char* segment_duration_check = 0;
const char* index = NULL;
char* tmp_index = NULL;
const char* http_prefix = NULL;
long max_tsfiles = NULL;
double prev_segment_time = 0;
double segment_duration = 0;
AVInputFormat* ifmt = NULL;
AVOutputFormat* ofmt = NULL;
AVFormatContext* ic = NULL;
AVFormatContext* oc = NULL;
AVStream* video_st = NULL;
AVStream* audio_st = NULL;
AVCodec* codec = NULL;
AVDictionary* pAVDictionary = NULL;
long frame_count = 0;
if (argc != 3) {
fprintf(stderr, "Usage: %s inputfile outputfile\n", argv[0]);
exit(1);
}
input = argv[1];
output = argv[2];
av_register_all();
char szError[256] = {0};
int nRet = avformat_open_input(&ic, input, ifmt, &pAVDictionary);
if (nRet != 0)
{
av_strerror(nRet, szError, 256);
printf(szError);
printf("\n");
printf("Call avformat_open_input function failed!\n");
return 0;
}
if (av_find_stream_info(ic) < 0)
{
printf("Call av_find_stream_info function failed!\n");
return 0;
}
ofmt = av_guess_format("mpegts", NULL, NULL);
if (!ofmt)
{
printf("Call av_guess_format function failed!\n");
return 0;
}
oc = avformat_alloc_context();
if (!oc)
{
printf("Call av_guess_format function failed!\n");
return 0;
}
oc->oformat = ofmt;
int video_index = -1, audio_index = -1;
for (unsigned int i = 0; i < ic->nb_streams && (video_index < 0 || audio_index < 0); i++)
{
switch (ic->streams[i]->codec->codec_type)
{
case AVMEDIA_TYPE_VIDEO:
video_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
video_st = add_output_stream(oc, ic->streams[i]);
break;
case AVMEDIA_TYPE_AUDIO:
audio_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
audio_st = add_output_stream(oc, ic->streams[i]);
break;
default:
ic->streams[i]->discard = AVDISCARD_ALL;
break;
}
}
codec = avcodec_find_decoder(video_st->codec->codec_id);
if (codec == NULL)
{
printf("Call avcodec_find_decoder function failed!\n");
return 0;
}
if (avcodec_open(video_st->codec, codec) < 0)
{
printf("Call avcodec_open function failed !\n");
return 0;
}
if (avio_open(&oc->pb, output, AVIO_FLAG_WRITE) < 0)
{
return 0;
}
if (avformat_write_header(oc, &pAVDictionary))
{
printf("Call avformat_write_header function failed.\n");
return 0;
}
//[[++
AVBitStreamFilterContext *bsfc = av_bitstream_filter_init("h264_mp4toannexb");
//AVBitStreamFilterContext *absfc = av_bitstream_filter_init("aac_adtstoasc");
if (!bsfc) {
fprintf(stderr, "bsf init error!\n");
return -1;
}
//]]++
int decode_done = 0;
do
{
double segment_time = 0;
AVPacket packet;
decode_done = av_read_frame(ic, &packet);
if (decode_done < 0)
break;
if (av_dup_packet(&packet) < 0)
{
printf("Call av_dup_packet function failed\n");
av_free_packet(&packet);
break;
}
//[[**
if (packet.stream_index == audio_index) {
segment_time = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
nRet = av_interleaved_write_frame(oc, &packet);
} else if (packet.stream_index == video_index) {
if (packet.flags & AV_PKT_FLAG_KEY) {
segment_time = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
} else {
segment_time = prev_segment_time;
}
//nRet = av_interleaved_write_frame(oc, &packet);
write_frame(oc, &packet, video_st->codec, bsfc);
}
//]]**
if (nRet < 0)
{
printf("Call av_interleaved_write_frame function failed: %d\n", nRet);
}
else if (nRet > 0)
{
printf("End of stream requested\n");
av_free_packet(&packet);
break;
}
av_free_packet(&packet);
frame_count++;
}while(!decode_done);
av_write_trailer(oc);
printf("frame_count = %d\n", frame_count);
av_bitstream_filter_close(bsfc);
avcodec_close(video_st->codec);
for(unsigned int k = 0; k < oc->nb_streams; k++)
{
av_freep(&oc->streams[k]->codec);
av_freep(&oc->streams[k]);
}
av_free(oc);
//getchar();
return 0;
}Compile this code, to got an executable file named
muxts, and then :$ ./muxts vid1.mp4 vid1.tsNo error message printed，but the audio stream was unsynchronized and noise。Check the .ts file using ffmpeg :
$ ffmpeg -i vid1.ts
ffmpeg version 0.8.14-tessus, Copyright (c) 2000-2013 the FFmpeg developers
built on Jul 29 2013 17:05:18 with llvm_gcc 4.2.1 (Based on Apple Inc. build 5658) (LLVM build 2336.1.00)
configuration: --prefix=/usr/local --arch=x86_64 --as=yasm --extra-version=tessus --enable-gpl --enable-nonfree --enable-version3 --disable-ffplay --enable-libvorbis --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-bzlib --enable-zlib --enable-postproc --enable-filters --enable-runtime-cpudetect --enable-debug=3 --disable-optimizations
libavutil 51. 9. 1 / 51. 9. 1
libavcodec 53. 8. 0 / 53. 8. 0
libavformat 53. 5. 0 / 53. 5. 0
libavdevice 53. 1. 1 / 53. 1. 1
libavfilter 2. 23. 0 / 2. 23. 0
libswscale 2. 0. 0 / 2. 0. 0
libpostproc 51. 2. 0 / 51. 2. 0
Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 90000.00 (180000/2)
Input #0, mpegts, from 'vid1.ts':
Duration: 00:00:03.75, start: 0.000000, bitrate: 3656 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0.0[0x100]: Video: h264 (Baseline), yuv420p, 640x480, 90k tbr, 90k tbn, 180k tbc
Stream #0.1[0x101]: Audio: aac, 48000 Hz, mono, s16, 190 kb/s
At least one output file must be specifiedWhat should i do ?
If this issue fixed , how can i concat multi .ts files into single .mp4 file ?
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During transcoding, does output quality of a video improve when i give output bitrate more than input video's bitrate ?
19 septembre 2013, par Jobin JoseI use ffmpeg for converting videos.
As i understand, the bitrate of a video stream is the number of bits which constitute the video over 1 second of time.
What happens when i specify the output video bitrate to be more than the input video's bitrate ?
For example :
If bitrate of "Input.mp4" is 2000KBps and i want to convert it to "Output.mp4" with output bitrate set to 3000KBps.
How will the converter create the extra 1000 bits(3000-2000) for every second of video ?
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